Re: [Asterisk-Users] Text to Speech - Someone needs to do this
People working on this have found that context influences the pronounciation of words. I think the root cause of this is that the human vocal tract cannot re-shape itself for different sounds instantly and must move from the previous sound to the next sound, we hear the movement. If it does instantly change then we hear it as un-natural robot-like speach. Your proposed system would sound just like what it is, a sequence of words. Good systems not only look at phonetic context but also inflection like tone, volume and pitch range and speed. Cursive hand writting is this way too. Cursive fonts don't look like real hand writting because each letter is always the same --- Matthew John Darnell [EMAIL PROTECTED] wrote: Why hasn't someone found 50 people who sound alike, put them in sound studios and record the 10,000 most commonly used words. You would all differnent forms of the 1,000 most words, i.e. leading, trailing, question etc. You can synthesize the other 0.05% when you run into them. With hard drives so big, processors so fast and EXT3 that can handle 30,000+ files in a single directory that seems like the way to do it. You could sell it for BIG bucks. -Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? SBC Yahoo! DSL - Now only $29.95 per month! http://sbc.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Poll - Would you pay $30-$50 for high quality speech synthesis?
Also almost forgot. They sell the demo voices on their site for 29.99. Linux and windows versions. Since I believe what they use is based off festival, perhaps the voices could be made to plug into the existing festival plugin for asterisk? I have been working with app_festival for about a week or so now trying to figure out what is going on with it... The existing app_festival has a serious bug in it that makes it unsuitable for production use. I posted about it before, but the jist is that the more channels that are trying to use app_festival at the same time, the more problems there are -- the channel will abort, and this will result in asterisk abandoning the call. I have tried tracing it with all sorts of things including stepping back through some truly monumental gdb logs. The problem is not on the festival side of things, as the preforking festival keeps up fine with asterisk, and the problem occurs even when using the festival cache (broken in the current code - my patch fixes it) I cannot find the bug after a week of poking at the code. As I intended to use app_festival as a temporary replacement for recorded voice prompts in an AGI application, it was no big deal at first as it works 100% if if you always wait for the speech to finish and only use one channel (fine for testing), but after having the flexibility to do speech synthesis, I can see that it would be a tremendously good application for even the IVR that I am working on... Anyway, I hope this speech project gets off the ground. Staying all OSS is very nice, but after spending a lot of time mucking with it, I'd be easily willing to spend $50 for a great sounding, working solution. I'm still going to be poking at app_festival, though, so if anyone has suggestions or understands some of the internals and wants to work on this with me, please mail me off-list. Having a 100% working and production-ready solution for festival would be good for a number of reasons and I'd like to see this happen too! ~John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GOTO inside AGI
I'm trying to Make a Goto inside a agi to another context/priority I used SET CONTEXT callh323, SET PRIORITY 1, SET EXTENSION s Apparently the SET EXTENSION is still assuming the value defined initially(), what is not defined in the new Context. Anyone has any turnaround for this? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Text to Speech - Someone needs to do this
I must say this is basically correct BUT Remember that festival is actually based phonetically. remember that and modify your text accordingly and you might be surprised at the results. yes the standard voices do suck ! On Tue, 15 Jul 2003 23:04:24 -0700 (PDT), Chris Albertson wrote: People working on this have found that context influences the pronounciation of words. I think the root cause of this is that the human vocal tract cannot re-shape itself for different sounds instantly and must move from the previous sound to the next sound, we hear the movement. If it does instantly change then we hear it as un-natural robot-like speach. Your proposed system would sound just like what it is, a sequence of words. Good systems not only look at phonetic context but also inflection like tone, volume and pitch range and speed. Cursive hand writting is this way too. Cursive fonts don't look like real hand writting because each letter is always the same --- Matthew John Darnell [EMAIL PROTECTED] wrote: Why hasn't someone found 50 people who sound alike, put them in sound studios and record the 10,000 most commonly used words. You would all differnent forms of the 1,000 most words, i.e. leading, trailing, question etc. You can synthesize the other 0.05% when you run into them. With hard drives so big, processors so fast and EXT3 that can handle 30,000+ files in a single directory that seems like the way to do it. You could sell it for BIG bucks. -Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? SBC Yahoo! DSL - Now only $29.95 per month! http://sbc.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phoneserve SIP provider
Lubomir Christov wrote:
yes
put something like this in your extension.conf
it will route all calls started with 0 (it will send the numbers without
0) to phoneserve accounts
exten = _0.,1,Dial(Sip/${EXTEN:[EMAIL PROTECTED],,)
exten = _0.,2,Dial(Sip/${EXTEN:[EMAIL PROTECTED],,)
Lubo
Thanks, I'll try this.
But will this automatically switch to the second channel if first is
busy?
Best reagrds,
Sergey Stasyuk
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phoneserve SIP provider
yes, just tray it :)
Sergey S. Stasyuk wrote:
Lubomir Christov wrote:
yes
put something like this in your extension.conf
it will route all calls started with 0 (it will send the numbers without
0) to phoneserve accounts
exten = _0.,1,Dial(Sip/${EXTEN:[EMAIL PROTECTED],,)
exten = _0.,2,Dial(Sip/${EXTEN:[EMAIL PROTECTED],,)
Lubo
Thanks, I'll try this.
But will this automatically switch to the second channel if first is
busy?
Best reagrds,
Sergey Stasyuk
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] g723.1 voicemail/conference files segfault *
Thanks Matteo,
Now I have a backtrace if that will help. I am not a programmer and this
really means nothing to me. I can only tell you that I have a g723.1 encoded
file (conf-onlyperson.g723) in /var/lib/asterisk/sounds/ when this happens.
#0 0x08058291 in ast_write (chan=0x8111718, fr=0x) at
channel.c:1332
1332switch(fr-frametype) {
(gdb) backtrace
#0 0x08058291 in ast_write (chan=0x8111718, fr=0x) at
channel.c:1332
#1 0x0805de8b in ast_readaudio_callback (data=0x8111718) at file.c:508
#2 0x0805f09d in ast_streamfile (chan=0x0, filename=0x41e42608
conf-onlyperson,
preflang=0x8111718 SIP/7600-894e) at file.c:575
#3 0x41e41733 in conf_run (chan=0x8111718, conf=0x80cd398, confflags=0) at
app_meetme.c:246
#4 0x41e40a73 in conf_exec (chan=0x8111718, data=0x8111718) at
app_meetme.c:585
#5 0x08060eca in pbx_exec (c=0x8111718, app=0x80f74a8, data=0x49dbcd2c,
newstack=1) at pbx.c:388
#6 0x08067ef8 in pbx_extension_helper (c=0x8111718, context=0x41e42608
conf-onlyperson,
exten=0x8111908 , priority=1, callerid=0x80e42e0 7600,
action=1105480556) at pbx.c:1130
#7 0x08062d2c in ast_pbx_run (c=0x41e44b6c) at pbx.c:1614
#8 0x080685b1 in pbx_thread (data=0x) at pbx.c:1830
#9 0x400252b6 in start_thread () from /lib/tls/libpthread.so.0
(gdb)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brancaleoni
Matteo
Sent: Tuesday, July 15, 2003 8:29 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] g723.1 voicemail/conference files segfault *
Off the topic: where can I find the core dump? I am running asterisk on
Redhat9.
in the dir where you started *. but you must have to issue
'ulimit -c unlimited' if you wanna asterisks dump cores.
if you're starting it via the init.astersik script, you will found
the cores in /tmp/
Matteo
--
Matteo Brancaleoni
Espia System Administrator
Email : [EMAIL PROTECTED]
Web : http://www.espia.it
Phone : +39.02.70633354 - ext 911
IAX(2): [EMAIL PROTECTED] - ext 911
or tel:17005662458 - ext 911
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960g
On Wed, 16 Jul 2003 [EMAIL PROTECTED] wrote: Has anybody tried Cisco 7960G? Or 7940? sure, using them all the time here (the Skinny version, which requires Cisco CallManager which in turn connects to asterisk via H.323). The hardware (same as for the SIP version, in fact you can convert between SIP, MGCP and Skinny versions by uploading new firmware) is pretty cool: great speaker phone, support for standard headset, big LCD. Just the handset is a bit chunky at least for European standards, and it often sounds worse than speakerphone mode... WRT the software, it's another thing, though: I found the phones crashing quite often, as soon as you try doing anything but standard phone calls (like using XML push services to display stuff on the LCD, or daring to assign non-CallManager URLs to some of the service buttons). So if stability is important, don't run them outside protected intranet environments! What audio compressions can I use with this phone and Asterisk? Reason why I'm asking is because Cisco supports G.711 and G.729a audio compression (probobaly some tohers but they are not listed on data sheet) and on Asterisk features i found that it supports G.729 but need licence. G.711 (both A-law and µ-Law) works fine with asterisk. (That's just log scale PCM audio, so while it comes at 64kbit/s, it's quite good quality.) What I'm asking is wheter Cisco 7960G is working with Asterisk and what can I expect from it (quality, codec support, ...) Sure is. I've seen people running the 79xx SIP models with Asterisk. G.711 is supported out of the Box whereas G.729 support for asterisk works if you purchase a license from Digium. Second question would be, are two SIP phones enough for testing/playing with Asterisk? Yes, but for fully functioning Music on Hold or conferences, you'll need a zaptel device: either one of the digium cards or one of the software dummies (ztdummy, zaprtc or hfcdummy, IIRC). They are required to provide timing interrupts for synchronization. Cheers, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] addmailbox2 (Attached)
Anyone wanting to go ahead and use VoiceMail2 will probably need this new version of the addmailbox utility.. I have updated the addmailbox utility to create the correct directory structure required by VM2 and copy the required files to the correct locations.. I just called it addmailbox2.. I don't know if it will be added to the CVS that is up to Mark.. Remember to make it executable.. :) Enjoy.. Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze addmailbox2 Description: Binary data
[Asterisk-Users] Cisco 7905G vs ATA186
Hi All, I'm looking at getting some Cisco VoIP hardware to play with in combination with a Asterisk server. I've heard that there is beta software available to do SIP on the 7905G. So, I'm thinking of either getting a 7905G or a ATA186. My dillema is, which one to buy? I can get both for about the same price, has anyone had any experience with using a 7905G with Asterisk? On one hand it would be useful to have a ATA186 for its two ports, might be useful for testing stuff (Can you call between the two ports on a ATA186 ok?). But on the other hand, having a proper IP Phone would be cool also. Cheers, Steven ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicemail instructions
Hi, I've been playing with Voicemail and Voicemail2 a bit for my users, and there are a few things I'm wondering about: - We can specify parameters to the mailbox (s, b or u) to select which prompts to play. However, if we specify 'b' or 'u' it plays that (customisable) message, but it also plays the voicemail instructions. For the dutch, it is customary that a user creates their own message which includes 'please leave your message after the tone' or similar, so the generic message is undesirable (or should be override-able). Is there something in the apps I've missed that allows this already ? - In voicemailmain2 there is no option in the menu that allows creating your own messages (in fact, option 3 is defunct). Is this in the coming, or am I missing more stuff ? Thanks! Met vriendelijke groet, Florian Overkamp ObSimRef BV (http://www.obsimref.com/) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Timeout in Call Transfering
hi, Can anybody pls tell me, how to increase the time gap between 2 digits when you transfer a call. ie, the operator answers the call, and presses hash key to transfer, and then enters the extension number, some times, it timeouts too quickly before the operator enters the whole extension number (may be bcos the operator is slow). I tried the following, but it doesn't seems to be helping when it comes to call transfering ... exten = s,4,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten = s,5,ResponseTimeout,10 ; Set Response Timeout to 10 seconds ... can anybody gv me an idea? Thank you inadvance, Surajee --This mail sent through OmniBIS.com-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail instructions
Hi, Use s before u or b together with Voicemail2 app. The default message will be skipped. BR, Dan P.S. I think it works with Voicemail too - Original Message - From: Florian Overkamp [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 16, 2003 2:56 PM Subject: [Asterisk-Users] voicemail instructions Hi, I've been playing with Voicemail and Voicemail2 a bit for my users, and there are a few things I'm wondering about: - We can specify parameters to the mailbox (s, b or u) to select which prompts to play. However, if we specify 'b' or 'u' it plays that (customisable) message, but it also plays the voicemail instructions. For the dutch, it is customary that a user creates their own message which includes 'please leave your message after the tone' or similar, so the generic message is undesirable (or should be override-able). Is there something in the apps I've missed that allows this already ? - In voicemailmain2 there is no option in the menu that allows creating your own messages (in fact, option 3 is defunct). Is this in the coming, or am I missing more stuff ? Thanks! Met vriendelijke groet, Florian Overkamp ObSimRef BV (http://www.obsimref.com/) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Web Based Frontend
Are there any web based frontends for asterisk, for mananging voice mail etc and asterisk in general? Kind Regards, Chris Bond ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7905G vs ATA186
As far as I know the Sip support for the 7905 has not been generally released so comments you've seen on this list refer to test versions of the code. You can set up a call between two phones on an ATA186 through asterisk. Iain --On Wednesday, July 16, 2003 9:28 pm +1000 Steven Honson [EMAIL PROTECTED] wrote: Hi All, I'm looking at getting some Cisco VoIP hardware to play with in combination with a Asterisk server. I've heard that there is beta software available to do SIP on the 7905G. So, I'm thinking of either getting a 7905G or a ATA186. My dillema is, which one to buy? I can get both for about the same price, has anyone had any experience with using a 7905G with Asterisk? On one hand it would be useful to have a ATA186 for its two ports, might be useful for testing stuff (Can you call between the two ports on a ATA186 ok?). But on the other hand, having a proper IP Phone would be cool also. Cheers, Steven ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail instructions
Hey Florian, On Wed, Jul 16, 2003 at 13:56:45 +0200, Florian Overkamp wrote: Hi, I've been playing with Voicemail and Voicemail2 a bit for my users, and there are a few things I'm wondering about: - We can specify parameters to the mailbox (s, b or u) to select which prompts to play. However, if we specify 'b' or 'u' it plays that (customisable) message, but it also plays the voicemail instructions. For the dutch, it is customary that a user creates their own message which includes 'please leave your message after the tone' or similar, so the generic message is undesirable (or should be override-able). Is there something in the apps I've missed that allows this already ? That's what the s is for. Use it together with the b or u to suppress the recording-instructions. This only works with Voicemail2, BTW, as the original Voicemail-app doesn't allow it to be used together with the other options. - In voicemailmain2 there is no option in the menu that allows creating your own messages (in fact, option 3 is defunct). Is this in the coming, or am I missing more stuff ? Mark talked about adding an extra menu some time ago and this was one of the features discussed, if I'm not mistaken. I'm not sure what the status is at the moment. Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P in Australia
Thanks for your reply. I tried changing the constant DEFAULT_CIDRINGS to 2. However I still don't get callerid information. Is this what you were referring to? Thanks, Hafeez On Wed, 16 Jul 2003 15:33:47 +1000, Gary [EMAIL PROTECTED] said: On Tue, 15 Jul 2003 17:44:56 -0800, hafeez bana wrote: G'Day, I need from regarding using the X100P. Some are specific to Australia and I have been told in IRC there are already existing users in oz, so I would appreciate your input. 1) Is the X100P Australia telco approved? NO 2) I have setup asterisk with X100P and whenever I dial in I cannot get any callerid information. Is this standard behaviour? If not how have other australian users managed to obtain the callerid information. in chan_zap.c it must be set for 2 rings not 2 for caller id 3) Volume and echo. The volume on the line is really low, there are also issues with the voice breaking up when I use an extension phone connected via the TDM40B and dial via the X100P. Is there any fix for this? The line is normally very clear. 4) What is ADSI? 5) Is there a site/book where I can learn about phone technology e.g E1, T1, Channel banks etc? 6) I have a friend in Kenya who I mentioned my digium experiments to and he is interested in purchasing the cards. However they use pulse phones back there and the lines are very unreliable. Does the digium hardware work with pulse dialling. Has anyone used digium hardware in countries where the phone infrastructure is bad? Thats it for now, Thanks, Hafeez -- hafeez bana [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- hafeez bana [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Text to Speech - Someone needs to do this
At 15:41 2003-07-15 -1000, Matthew John Darnell wrote: Why hasn't someone found 50 people who sound alike, put them in sound studios and record the 10,000 most commonly used words. You would all differnent forms of the 1,000 most words, i.e. leading, trailing, question etc. You can synthesize the other 0.05% when you run into them. With hard drives so big, processors so fast and EXT3 that can handle 30,000+ files in a single directory that seems like the way to do it. You could sell it for BIG bucks. Text-to-Speech (TTS) is usually either formative, created by synthesis of sounds; or concatenative, created by concatenating sounds of actual speech samples. However, concatenative TTS usually works by using small fragments of speech, not entire words. The storage requirements are much smaller, and it gives the system an opportunity to pick units of speech that match the units of speech that precede and follow them. The real trick is to get the correct posidy. Here's three sentences with the same words but each with different prosidy: I said 'yes.' I said yes? _I_ said '_yes_'???!! Both formative and concatenative systems add prosidy. Adding prosidy to whole-word concatentative systems is difficult. If you're in a buying mood, there are some excellent TTS systems available. For example, Rhetorical (http://www.rhetorical.com) has some excellent voices. And they have the funniest TTS current available is the Southern California female voice; I use it for non-serious demos (That's so totally awesome.) Commercial TTS is actually very intelligble and perfectly adequate for many tasks. -- Moshe Yudkowsky Disaggregate 2952 W Fargo Chicago, IL 60645 USA www.Disaggregate.com [EMAIL PROTECTED] +1 773 764 8727 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Cygwin?
Is the hassle in running it or setting it up? This gets back to my interest in a CD to boot and install a basic system on a hard drive. Something like a 2 line 4 station version and then a single T1, 4 station, 2 line. This is why there is a users list and a developers list. As a user, I just want a CD with typical config ready to go. As a developer, I want to play with and tweak everything, including the OS. Others are asking for a GUI or web interface. There's a place for all of this. Look at what's involved in getting started: You either have to download 600+MB Linux, install, compile, etc. Or run out and buy a Linux version and install, compile, etc. Now for most of us this is not a big problem. But, just look at the time envolved in setting up a couple of 266mz boxes to play and test with. Put an ISO on the site and watch hardware sales fly... And then watch the consultants market grow. There will be posts like: ...well I bought the hardware, installed, it works but I need xxxyyy, can any one log into my system and program this thing?... James Taylor [EMAIL PROTECTED] 903-793-1953 -- Original Message -- From: Chris Earle \(CBL\) [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Wed, 16 Jul 2003 02:23:12 -0400 Hey all, quick question: does asterisk work okay in a Cygwin environment? I want to install it on my cygwin setup for local testing/demoing and save me the hassle of using a pure linux machine As long as it doesn't take a huge huge performance hit from running out of Cygwin, then I'll have a go there for a start confirmation appreciated! thanks -- C h r i sE a r l e System Solutions Specialist ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- James Taylor [EMAIL PROTECTED] 903-793-1953 -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail instructions
Hi, Me too, but look at a similar thread started by me in the message archive..:-) BR, Dan - Original Message - From: Florian Overkamp [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 16, 2003 4:45 PM Subject: Re: [Asterisk-Users] voicemail instructions Hi, At 15:16 16-7-2003 +0300, you wrote: Use s before u or b together with Voicemail2 app. The default message will be skipped. Hmm, I was put off by the At most one of 's', 'u', or 'b' may be specified. in the help lines... (Thanks, Traveller too) Met vriendelijke groet, Florian Overkamp ObSimRef BV (http://www.obsimref.com/) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Vendors for phones
I'm in the process of setting up a test/demonstration system to show that VoIP is realistic and applicable for our needs. We put a 7905 and 7960 on a request for quote that went out the other day (to people like CDW Microwarehouse). All of the vendors returned thier quotes without including the Cisco phones. So my question: where do you buy your phones? We can't buy direct from Cisco (must have 3 quotes). Thanks... ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream sip phone
I asked [EMAIL PROTECTED] the other day. They wrote back: US list retail price of BudgeTone SIP phones: Model 101 $75/ea (available now) Model 102 $85/ea (available now) US list retail price of HandyTone VoIP analog telephone adaptor: $75/ea (available in late July 2003) Please contact our reseller (Ovislink/dgtimes) regarding your sample purchase. James @ Ovislink/dgtimes can be reached at tel: (626) 854-1805 or fax: 626.854.0835 and [EMAIL PROTECTED] Their web site is at: www.ovislink.com On Wed, 16 Jul 2003, Marian Danisek wrote: hello, i found in list archives some notes about grandstream sip voip phones. Does anybody succesfuly tested those phones with asterisk ? Mark ? What about the prices ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Cygwin?
On Wed, 2003-07-16 at 09:10, jltaylor wrote: Is the hassle in running it or setting it up? This gets back to my interest in a CD to boot and install a basic system on a hard drive. Something like a 2 line 4 station version and then a single T1, 4 station, 2 line. This is why there is a users list and a developers list. As a user, I just want a CD with typical config ready to go. As a developer, I want to play with and tweak everything, including the OS. But there is _NO_ typical config. This is enough of a problem. Plus there is no need to host an ISO of the OS and cost digium money in bandwidth that other people are more than willing to do. The maintaining of a OS ISO is immense and best left to other projects. In fact the only thing really needed is for someone to set up a nightly build and package of asterisk into the couple of different package formats and make it available to the world. Others are asking for a GUI or web interface. There's a place for all of this. Look at what's involved in getting started: You either have to download 600+MB Linux, install, compile, etc. Or run out and buy a Linux version and install, compile, etc. Now for most of us this is not a big problem. But, just look at the time envolved in setting up a couple of 266mz boxes to play and test with. See you need to learn about other distros and installers. I know that Mandrake and RH offer network installs, and debian shouldn't be installed any other way. I'm only commenting on debians network install because I know it, but you only download 28 megs of files and then only what you are going to install after that. Total download for an asterisk machine should be under 150 megs. Put an ISO on the site and watch hardware sales fly... Do you think the ISO will change all these VoIP only users into hardware users? If you listen to the comments from them, it is a cost issue mostly on the hardware, not the software. No amount of software bundling is going to change the budget of a user. And then watch the consultants market grow. There will be posts like: ...well I bought the hardware, installed, it works but I need xxxyyy, can any one log into my system and program this thing?... I doubt this. The consultants market will be more of the kind like VCCH is doing which is going out to a site and saying, We can provide you this, that, and these other things all for a price under that quote you have in your hands now. The difference here is that most users that already found their way here and went ahead with a purchase of hardware will either already know how to do it themselves, or are patient enough to wait till that feature comes forward. Those who need consultants usually will not be the ones we see. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream sip phone
On Wed, 2003-07-16 at 15:44, Marian Danisek wrote: hello, i found in list archives some notes about grandstream sip voip phones. Does anybody succesfuly tested those phones with asterisk ? Mark ? They seem to work with asterisk. I don't yet have a couple myself but on irc there are people who use them. Join #asterisk on irc.freenode.net (or .com don't remember) and ask around. What about the prices ? $85 for the 102 regards Marian Regards, Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dial SessionTime
In cdr table or in /var/log/asterisk/cdr-csv/Master.csv srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de [EMAIL PROTECTED] Enviado el: miércoles, 16 de julio de 2003 23:54 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] Dial SessionTime Hi Folks, After a successful Dial/H323, is there any to get the conversation duration time? Thanks, Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream sip phone
I have been testing a couple of them for about 2 weeks now.. They are very good for the price.. The only issue that I still have is that the phone does not seem to be able to pickup the time correctly from an NTP server that is not on the local network so the display always shows 1900-XX-XX for the date.. This issue I am sure will be solved in the near future.. I also have SNOM200's which are awesome phones but they are over twice the price including shipping of the GS phones to the UK.. Without shipping I would be able to get nearly 3 GS phones for the price of one SNOM200... Unfortunately the GS phone does not have a GSM codec but it does support just about every other codec out there and supports just about every feature you could want for a standard desktop phone.. If you want to know more let me know.. hello, i found in list archives some notes about grandstream sip voip phones. Does anybody succesfuly tested those phones with asterisk ? Mark ? What about the prices ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip codec preferences
Hi. I'm experiencing a issue (not big, but important) I have an asterisk installation with a buch of sip phones analog ones. I have 2 1 sip phone that's outside in the world, and is nat'ed. I'm using g.729 with it. I wanna use g.729 only for the remote phone, and ulaw for the local ones, since they're on a lan. What happens? when I call the remote phone, g.729 is used, but when the remote calls ulaw is used... beside there's a disallow=all , allow=g729 is the user definition. so seems that when we call it, the codec definitions are taken from the user config itself, but when it call us, the codec defs are from the global settings. that's the same if we call remote (or receive) from an analog or iax phone. Here's a snippet of my sip.conf: ; ; SIP Configuration for Asterisk ; [general] port = 5060 bindaddr = 0.0.0.0 context = local tos = lowdelay disallow = all allow = ulaw ;local phone definition [200] accountcode=localphone mailbox=200 type=friend secret=secret username=200 host=dynamic callgroup=1 pickupgroup=1 ; remote phone definition [250] accountcode=remotephone type=friend secret=X nat=yes username=250 context=local reinvite=no disallow=all allow=g729 canreinvite=no host=dynamic qualify=1000 callgroup=1 pickupgroup=1 Any hint? -- Matteo Brancaleoni Powered by RedHat Linux 8.0 Linux User #153521 -BEGIN GEEK CODE BLOCK- Version: 3.12 GS d? s:- a- C+++ UL P+ L+++ E- W+++ N++ o K- w-- O- M-- V-- PS PE- Y PGP++ t 5 X+ R tv- b++ DI D+ G e h! r++ y --END GEEK CODE BLOCK-- -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39.02.70633354 - ext 911 IAX(2): [EMAIL PROTECTED] - ext 911 or tel:17005662458 - ext 911 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P in Australia (was Asterisk-Users digest, Vol 1 #840 - 13 msgs)
From: hafeez bana [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Wed, 16 Jul 2003 05:17:59 -0800 Subject: Re: [Asterisk-Users] X100P in Australia Reply-To: [EMAIL PROTECTED] Thanks for your reply. I tried changing the constant DEFAULT_CIDRINGS to 2. However I still don't get callerid information. Is this what you were referring to? Thanks, Hafeez You don't have another caller ID device on the line do you? My X100P cards won't recognise the caller ID when my cordless phone is plugged into the line (which also has caller ID). Also, I haven't been monitoring the list too much; so I don't know if this has been mentioned; but do you have the following in your zapata.conf: usecallerid=yes callerid=asreceived Once that was done on my system; the X100P cards pickup caller ID fine for me (I'm in Sydney). -Shaun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Analog features over the ATA-186
I have been using the ATA-186 with good success (with exception of the fact that you have to recycle it from time to time). The one thing I havent been able to do is to figure out how to make use of parking, transfers, etc. My cordless doesnt seem to pass DTMF, so I havent had any success with parking. As for transfers, I can do a flash and get dial tone, but then what are my options at that point for transferring? Any help would be greatly appreciated! Kim C. Callis
Re: [Asterisk-Users] voicemail instructions
fixed in CVS thanks! On Wed, 16 Jul 2003, Florian Overkamp wrote: Hi, At 15:16 16-7-2003 +0300, you wrote: Use s before u or b together with Voicemail2 app. The default message will be skipped. Hmm, I was put off by the At most one of 's', 'u', or 'b' may be specified. in the help lines... (Thanks, Traveller too) Met vriendelijke groet, Florian Overkamp ObSimRef BV (http://www.obsimref.com/) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Cygwin?
Thanks for your enthuastic response. There's this Linux project out there for 802.11 at: www.station-server.com They have figured out how to make this type of distribution package work. Don't get me wrong, Asterisk seems to have just about everything from a feature standpoint. The open source concept is one that I support. We run FreeBSD for routers and I love it. You are absolutely correct about the need to learn about Linux distributions and installers. Most people don't and some find it too difficult (I suppose that they are the ones who should stick to Windows?). Hardware costs? I guess these guys that have a hardware cost problem have never priced a Dialogic 240xx/T1 or the quad card. Used single T1 Dialogic cards are $1100. *** The digium hardware offerings are the best price that I've seen for any solution. *** -- Original Message -- From: Steven Critchfield [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: 16 Jul 2003 09:14:06 -0500 On Wed, 2003-07-16 at 09:10, jltaylor wrote: Is the hassle in running it or setting it up? This gets back to my interest in a CD to boot and install a basic system on a hard drive. Something like a 2 line 4 station version and then a single T1, 4 station, 2 line. This is why there is a users list and a developers list. As a user, I just want a CD with typical config ready to go. As a developer, I want to play with and tweak everything, including the OS. But there is _NO_ typical config. This is enough of a problem. Plus there is no need to host an ISO of the OS and cost digium money in bandwidth that other people are more than willing to do. The maintaining of a OS ISO is immense and best left to other projects. In fact the only thing really needed is for someone to set up a nightly build and package of asterisk into the couple of different package formats and make it available to the world. Others are asking for a GUI or web interface. There's a place for all of this. Look at what's involved in getting started: You either have to download 600+MB Linux, install, compile, etc. Or run out and buy a Linux version and install, compile, etc. Now for most of us this is not a big problem. But, just look at the time envolved in setting up a couple of 266mz boxes to play and test with. See you need to learn about other distros and installers. I know that Mandrake and RH offer network installs, and debian shouldn't be installed any other way. I'm only commenting on debians network install because I know it, but you only download 28 megs of files and then only what you are going to install after that. Total download for an asterisk machine should be under 150 megs. Put an ISO on the site and watch hardware sales fly... Do you think the ISO will change all these VoIP only users into hardware users? If you listen to the comments from them, it is a cost issue mostly on the hardware, not the software. No amount of software bundling is going to change the budget of a user. And then watch the consultants market grow. There will be posts like: ...well I bought the hardware, installed, it works but I need xxxyyy, can any one log into my system and program this thing?... I doubt this. The consultants market will be more of the kind like VCCH is doing which is going out to a site and saying, We can provide you this, that, and these other things all for a price under that quote you have in your hands now. The difference here is that most users that already found their way here and went ahead with a purchase of hardware will either already know how to do it themselves, or are patient enough to wait till that feature comes forward. Those who need consultants usually will not be the ones we see. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- James Taylor [EMAIL PROTECTED] 903-793-1953 -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Cygwin?
On Wed, 2003-07-16 at 11:47, jltaylor wrote: Thanks for your enthuastic response. There's this Linux project out there for 802.11 at: www.station-server.com They have figured out how to make this type of distribution package work. And there is nothing stopping you from getting a knoppix CD started that will do the same sort of thing. Don't get me wrong, Asterisk seems to have just about everything from a feature standpoint. The open source concept is one that I support. We run FreeBSD for routers and I love it. You are absolutely correct about the need to learn about Linux distributions and installers. Most people don't and some find it too difficult (I suppose that they are the ones who should stick to Windows?). I tried my best to not include the windows comments in my post. It wasn't a route I wanted to go in this argument. But since we are here anyways, I'm concerned that if it becomes too easy to have a system connected to the PSTN and the net that we will see people looking this way for exploits. The difficulty here is that they don't have to exploit asterisk to get in and set up new accounts and then use asterisk to route them to the PSTN. The current level of knowledge required to get started should help weed out the people who helped NIMDA and CODE_RED along. We all know they exist no matter what OS they use. Hardware costs? I guess these guys that have a hardware cost problem have never priced a Dialogic 240xx/T1 or the quad card. Used single T1 Dialogic cards are $1100. *** The digium hardware offerings are the best price that I've seen for any solution. *** And that is the very reason why my company and myself personally have bought hardware from Digium. We currently have 4 machines with 320 cards that seem to go for over $1500 used, and it is only 16 analog FXO ports. -- Original Message -- From: Steven Critchfield [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: 16 Jul 2003 09:14:06 -0500 On Wed, 2003-07-16 at 09:10, jltaylor wrote: Is the hassle in running it or setting it up? This gets back to my interest in a CD to boot and install a basic system on a hard drive. Something like a 2 line 4 station version and then a single T1, 4 station, 2 line. This is why there is a users list and a developers list. As a user, I just want a CD with typical config ready to go. As a developer, I want to play with and tweak everything, including the OS. But there is _NO_ typical config. This is enough of a problem. Plus there is no need to host an ISO of the OS and cost digium money in bandwidth that other people are more than willing to do. The maintaining of a OS ISO is immense and best left to other projects. In fact the only thing really needed is for someone to set up a nightly build and package of asterisk into the couple of different package formats and make it available to the world. Others are asking for a GUI or web interface. There's a place for all of this. Look at what's involved in getting started: You either have to download 600+MB Linux, install, compile, etc. Or run out and buy a Linux version and install, compile, etc. Now for most of us this is not a big problem. But, just look at the time envolved in setting up a couple of 266mz boxes to play and test with. See you need to learn about other distros and installers. I know that Mandrake and RH offer network installs, and debian shouldn't be installed any other way. I'm only commenting on debians network install because I know it, but you only download 28 megs of files and then only what you are going to install after that. Total download for an asterisk machine should be under 150 megs. Put an ISO on the site and watch hardware sales fly... Do you think the ISO will change all these VoIP only users into hardware users? If you listen to the comments from them, it is a cost issue mostly on the hardware, not the software. No amount of software bundling is going to change the budget of a user. And then watch the consultants market grow. There will be posts like: ...well I bought the hardware, installed, it works but I need xxxyyy, can any one log into my system and program this thing?... I doubt this. The consultants market will be more of the kind like VCCH is doing which is going out to a site and saying, We can provide you this, that, and these other things all for a price under that quote you have in your hands now. The difference here is that most users that already found their way here and went ahead with a purchase of hardware will either already know how to do it themselves, or are patient enough to wait till that feature comes forward. Those who need consultants usually will not be the ones we see. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED]
Re: [Asterisk-Users] Enhanced queue app
Mark, Any news on the enhanced queue app progress? Just wondering. Jim Friedeck ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Text to Speech - Someone needs to do this
--- Moshe Yudkowsky [EMAIL PROTECTED] wrote: SNIP The real trick is to get the correct posidy. Here's three sentences with the same words but each with different prosidy: I said 'yes.' I said yes? _I_ said '_yes_'???!! Both formative and concatenative systems add prosidy. Adding prosidy to whole-word concatentative systems is difficult. The thing is that _people_ don't do text to speech. If you were to simply read one word at a time you'd sound bad too. Try it: if, ... you. ...were, ... to, ... simply, ...read, ... You sound like a robot. No, we people know what it is we are trying to comunicate if you want a synthetic voice to sound natural you will have to tell the software the _intent_ of the words not just the words. You would need a markup language for that emph I /emph said quotequestionword yes /quote/questionword now the system can apply some transformations to the pitch, speed and loudness. For interactive systems markup works because the software generating the text knows _why_ it is generating the text Reading a book for the blind is a much harder problem. The TTS system has to do the same job as a voice actor which even includes understands the emotions of characters in a novel. Very hard to do for a computer. But interactive systems can use markup to get the expresson right. And don't put down festival. Many (most?) of the comercial systems _are_ festival. you, = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? SBC Yahoo! DSL - Now only $29.95 per month! http://sbc.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiple Phones for 1 Extension
Hi, I'd like to have a SIP phone at home and at the office and have them both ring when my extension is dialed. Right now I used the same config for the phones (Cisco 7960's). So they both register with the same login pw. This doesn't seem to work quiet right, where only the last phone to register seems to get the calls. What is the proper way to set this up? Thanks, Justin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multiple Phones for 1 Extension
I'd like to have a SIP phone at home and at the office and have them both ring when my extension is dialed. Right now I used the same config for the phones (Cisco 7960's). So they both register with the same login pw. This doesn't seem to work quiet right, where only the last phone to register seems to get the calls. What is the proper way to set this up? Have the phones register with different names (make a separate entry for each phone in sip.conf) then specify them both in the dial string separated by '' .. you can specify as many as you want and all will ring. The one to answer gets the call, naturally. exten = 3000,1,Dial(SIP/phoneoneSIP/phonetwo) John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple Phones for 1 Extension
On Wed, 2003-07-16 at 12:20, Justin Eckhouse wrote: Hi, I'd like to have a SIP phone at home and at the office and have them both ring when my extension is dialed. Right now I used the same config for the phones (Cisco 7960's). So they both register with the same login pw. This doesn't seem to work quiet right, where only the last phone to register seems to get the calls. What is the proper way to set this up? Multiple usernames and passwords, then use the dial command properly -- show application dial -- -= Info about application 'Dial' =- [Synopsis]: Place an call and connect to the current channel [Description]: Dial(Technology/resource[Technology2/resource2...][|timeout][|options][|URL]): Requests one or more channels and places specified outgoing calls on them. From this you should not that you could do... exten = s,1,Dial(SIP/workSIP/home) -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Text to Speech - Someone needs to do this
At 10:11 2003-07-16 -0700, Chris Albertson wrote: SNIP if you want a synthetic voice to sound natural you will have to tell the software the _intent_ of the words not just the words. You would need a markup language for that emph I /emph said quotequestionword yes /quote/questionword The W3C has a TTS markup language, SSML, http://www.w3.org/TR/speech-synthesis/. However, SSML is not a _semantic_ markup language. SSML gives directives about prosidy and pronunciation. And don't put down festival. Many (most?) of the comercial systems _are_ festival. I am not putting down Festival. However, I don't believe that many or most commercial systems are based on Festival. I think we should take any further discussion off-list. Regards, Moshe -- Moshe Yudkowsky Disaggregate 2952 W Fargo Chicago, IL 60645 USA http://www.Disaggregate.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Text to Speech - Someone needs to do this
Moshe Yudkowsky wrote: At 10:11 2003-07-16 -0700, Chris Albertson wrote: SNIP if you want a synthetic voice to sound natural you will have to tell the software the _intent_ of the words not just the words. You would need a markup language for that emph I /emph said quotequestionword yes /quote/questionword The W3C has a TTS markup language, SSML, http://www.w3.org/TR/speech-synthesis/. However, SSML is not a _semantic_ markup language. SSML gives directives about prosidy and pronunciation. Two interesting things about SSML (which used to be called Sable). One - there is almost no support for it amongst the commercial TTS packages. Two - even the people who wrote the SSML spec don't seem to have fully implemented it. The markup in most commercial TTS software is both proprietary and cranky. And don't put down festival. Many (most?) of the comercial systems _are_ festival. I am not putting down Festival. However, I don't believe that many or most commercial systems are based on Festival. You are wrong. All the packages I know, except Eloquence and maybe RealSpeak, are based at some level on Festival. The ones derived from Naturally Speaking have most of the Festival directories still in place. Strange, but true. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems getting 7960's to play nice with Asterisk
I'm trying to get my newly flashed 7960's to play nice with Asterisk, but I'm having some problems. I can get my 7960 to register with the proxy, and if I dial my own extension, according to my dial plan asterisk should transfer me to voice mail. Asterisk thinks its playing me voice mail prompts, but the phone just rings busy. I'm using the 4.4 cisco flash. Other things: The phone constantly says Ethernet Disconnected. Even though it tftp's configs and registers with the proxy. I get no dialtone on the SIP phone. I'm using the default diaplan.xml file. I'm running asterisk from cvs as of yesterday. I have one zaptel interface but I haven't configured * for it yet. (I figured being able to call voicemail was a good first test). Any help would be greatly appreciated! here is my sip.cnf: [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 192.168.3.5; Address to bind to (all addresses on machine) allow=all ; Allow all codecs context = bogon-calls ; Send SIP callers that we don't know about here [2000] type=friend ; This device takes and makes calls username=2000 ; Username on device secret=phone1 ; Password for device host=dynamic ; This host is not on the same IP addr every time context=from-sip ; Inbound calls from this host go here mailbox=100 ; Activate the message waiting light if this ; voicemailbox has messages in it dtmfmode=rfc2833 ;canreinvite=no [2001]; Duplicate of 2000, except with different auth data type=friend username=2001 secret=2001test host=dynamic context=from-sip More info: From tethereal dump: 70.822347 192.168.3.126 - 192.168.3.5 SIP Request: REGISTER sip:192.168.3.5 70.823464 192.168.3.5 - 192.168.3.126 SIP Status: 100 Trying 70.823547 192.168.3.5 - 192.168.3.126 SIP Status: 407 Proxy Authentication Required 70.860296 192.168.3.126 - 192.168.3.5 SIP Request: REGISTER sip:192.168.3.5 70.861228 192.168.3.5 - 192.168.3.126 SIP Status: 100 Trying 70.861298 192.168.3.5 - 192.168.3.126 SIP Status: 407 Proxy Authentication Required 70.970580 192.168.3.126 - 192.168.3.5 SIP Request: REGISTER sip:192.168.3.5 70.970945 192.168.3.5 - 192.168.3.126 SIP Status: 100 Trying 70.971214 192.168.3.5 - 192.168.3.126 SIP Status: 200 OK 71.088456 192.168.3.126 - 192.168.3.5 SIP Request: REGISTER sip:192.168.3.5 71.088752 192.168.3.5 - 192.168.3.126 SIP Status: 100 Trying 71.088940 192.168.3.5 - 192.168.3.126 SIP Status: 200 OK 75.108124 192.168.3.5 - 192.168.3.126 SIP/SDP Request: NOTIFY sip:[EMAIL PROTECTED], with session description 75.133604 192.168.3.126 - 192.168.3.5 SIP Status: 200 OK 82.392427 192.168.3.126 - 192.168.3.5 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED];user=phone, with session description 82.393494 192.168.3.5 - 192.168.3.126 SIP Status: 407 Proxy Authentication Required 82.448428 192.168.3.126 - 192.168.3.5 SIP Request: ACK sip:[EMAIL PROTECTED];user=phone 82.493497 192.168.3.126 - 192.168.3.5 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED];user=phone, with session description 82.494235 192.168.3.5 - 192.168.3.126 SIP Status: 100 Trying 82.495603 192.168.3.5 - 192.168.3.126 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 82.621388 192.168.3.126 - 192.168.3.5 SIP Status: 500 Internal Server Error 82.621816 192.168.3.5 - 192.168.3.126 SIP Request: ACK sip:[EMAIL PROTECTED] 82.622283 192.168.3.5 - 192.168.3.126 SIP/SDP Status: 200 OK, with session description 82.751118 192.168.3.126 - 192.168.3.5 SIP Request: BYE sip:[EMAIL PROTECTED];user=phone 82.751681 192.168.3.5 - 192.168.3.126 SIP Status: 200 OK 83.629324 192.168.3.5 - 192.168.3.126 SIP/SDP Status: 200 OK, with session description From Asterisk console: -- Registered SIP '2000' at 192.168.3.126 port 5060 expires 3600 -- Executing Dial(SIP/2000-6000, SIP/2000|20) in new stack -- Called 2000 -- Got SIP response 500 Internal Server Error back from 192.168.3.126 -- SIP/2000-6dcd is circuit-busy == Everyone is busy at this time -- Executing VoiceMail(SIP/2000-6000, b2000) in new stack == Parsing '/etc/asterisk/voicemail.conf': Found -- Playing 'vm/2000/busy' == Spawn extension (from-sip, 2000, 102) exited non-zero on 'SIP/2000-6000' WARNING[196621]: File chan_sip.c, Line 415 (retrans_pkt): Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Response) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX pauses
Hi, I'm running asterisk in the following setup Phone - WX100USB - * - Internet - * - WX100P - PSTN The two Asterisks talk to each other via IAX2 and use GSM for voice. This seems to work quite well except for occasional pauses in voice transmission. These seem to occur in _one_ direction only (when I'm on the phone, I can't hear the person that I called via the PSTN), last several seconds (as in one to five seconds) and are unrelated to network connectivity (a ping in another window runs just fine all the time). What could be the cause? What else could I do to help hunt down that bug? --J. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream sip phone
I'm working greatly with 40+ Grandstream phones. Audio quality is good enough for production environment, the cost is really low and the configuration is *Really* easy. But a little answer to Wipeout is: The only issue that I still have is that the phone does not seem to be able to pickup the time correctly from an NTP server that is not on the local network so the display always shows 1900-XX-XX for the date.. This issue I am sure will be solved in the near future.. Have you tried to mantain the default ntp server on your phone? (the *.gov one) I normally use internal ntp servers but in a particular context i've used that ntp server and it worked perfectly. Could be a Firewall issue, maybe? It works on every firmware since .58, for me. -- Stefano ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX pauses
Hey Jan, On Wed, Jul 16, 2003 at 11:45:13 -0700, Jan Rychter wrote: Hi, I'm running asterisk in the following setup Phone - WX100USB - * - Internet - * - WX100P - PSTN The two Asterisks talk to each other via IAX2 and use GSM for voice. This seems to work quite well except for occasional pauses in voice transmission. These seem to occur in _one_ direction only (when I'm on the phone, I can't hear the person that I called via the PSTN), last several seconds (as in one to five seconds) and are unrelated to network connectivity (a ping in another window runs just fine all the time). What could be the cause? What else could I do to help hunt down that bug? I have the exact same problem over here, and it seems some others on the list and on IRC have it as well. I'm still using IAX1 because of it. My setup is very similar to yours, except that I also use SIP and H.323- phones locally, which doesn't seem to make any difference. Both Asterisk-boxes have Zaptel-hardware in them (which seems to make a difference, as IAX2 uses it's timing, if available), I use GSM and trunking is off (turning it on didn't solve this problem, though). Grtz, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX pauses
turn off jitterbuffer in both servers. aka jitterbuffer=no in iax.conf jitterbuffer, unfortunately, is buggy and don't work as expected. Il mer, 2003-07-16 alle 20:45, Jan Rychter ha scritto: Hi, I'm running asterisk in the following setup Phone - WX100USB - * - Internet - * - WX100P - PSTN The two Asterisks talk to each other via IAX2 and use GSM for voice. This seems to work quite well except for occasional pauses in voice transmission. These seem to occur in _one_ direction only (when I'm on the phone, I can't hear the person that I called via the PSTN), last several seconds (as in one to five seconds) and are unrelated to network connectivity (a ping in another window runs just fine all the time). What could be the cause? What else could I do to help hunt down that bug? --J. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39.02.70633354 - ext 911 IAX(2): [EMAIL PROTECTED] - ext 911 or tel:17005662458 - ext 911 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7910 compatibility
Hello, I'm looking at getting some Cisco 7910G+SW phones through a local liquidator. Is anyone currently using these with asterisk? Any special limitations with this one as compared to the 7940 and 7960's? Will they work with SIP or are they stuck in Cisco-land with Call-Manager only? Thanks, MATT--- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream sip phone
On Wednesday 16 July 2003 03:52 pm, Greg Renouf wrote: Grandstream can improve the quality of their 'user interface' (many others have already accomplished this goal,) I can see very few situations where the $10-20 cost saving will make the quality sacrifice worthwhile. What other phones are in the $95-$105 range ??? Regards...Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream sip phone
Have you tried to mantain the default ntp server on your phone? (the *.gov one) I normally use internal ntp servers but in a particular context i've used that ntp server and it worked perfectly. I have tried many public NTP servers and all have the same result.. Could be a Firewall issue, maybe? No its not the firewall becasue I have no problems setting time on variout PC's using NTP and public servers.. Thanks for the thoughts anyway.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream sip phone
- Original Message - From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 16, 2003 10:19 PM Subject: Re: [Asterisk-Users] grandstream sip phone I have tried many public NTP servers and all have the same result.. Wait. I have tried many public ntp too. It worked ONLY with the default one. Check for this if you haven't done yet. At the present I don't remember (Sorry) the exact server which worked for me, since I've installed an ntp service on my server. -- Stefano ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX pauses
In iax.conf, do you have jitterbuffer set, try jitterbuffer=no, or try some values from 1-5, i use 3 -Original Message- From: Jan Rychter [EMAIL PROTECTED] To: [EMAIL PROTECTED] [EMAIL PROTECTED] Date: July 16, 2003 11:45 AM Subject: [Asterisk-Users] IAX pauses Hi, I'm running asterisk in the following setup Phone - WX100USB - * - Internet - * - WX100P - PSTN The two Asterisks talk to each other via IAX2 and use GSM for voice. This seems to work quite well except for occasional pauses in voice transmission. These seem to occur in _one_ direction only (when I'm on the phone, I can't hear the person that I called via the PSTN), last several seconds (as in one to five seconds) and are unrelated to network connectivity (a ping in another window runs just fine all the time). What could be the cause? What else could I do to help hunt down that bug? --J. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems getting 7960's to play nice with Asterisk
Other things: The phone constantly says Ethernet Disconnected. Even though it tftp's configs and registers with the proxy. Something is wrong with either the phone hardware itself, the network port on the hub or switch it is attached to, the ethernet cable it is connected with, or the 10/100 autonegotiation between the phone and the switch. Any other problems you are having are quite likely due to these network problems, and so you should fix this problem before you attempt to debug any of the others. The Ethernet Disconnected message on the phones is the equivalent of having no link light on your NIC. Standard network connection debugging applies. Some tips: This problem is likely due to a bad cable or 10/100 auto negotiation issues. Try known, working cables on a known working switch/hub port to see if the problem occurs. Try plugging it into another switch (preferably of different brand) if you have one. The 7960 cannot be forced to either 10 or 100mb operation, so if you do determine that it's not the cable or phone hardware, you'll have to either find a way to force 10 or 100mb operation on your switch or you'll have to get a new one... Once you get your phone hooked up to the network without problems, then you can start looking at the software issues.. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems getting 7960's to play nice withAsterisk
This phone was working with the same cable/switch when used with a Skinny configuration. But I'll try your suggestion out just to make sure when I get home tonight... -Steve On Wed, 16 Jul 2003, John Laur wrote: Other things: The phone constantly says Ethernet Disconnected. Even though it tftp's configs and registers with the proxy. Something is wrong with either the phone hardware itself, the network port on the hub or switch it is attached to, the ethernet cable it is connected with, or the 10/100 autonegotiation between the phone and the switch. Any other problems you are having are quite likely due to these network problems, and so you should fix this problem before you attempt to debug any of the others. The Ethernet Disconnected message on the phones is the equivalent of having no link light on your NIC. Standard network connection debugging applies. Some tips: This problem is likely due to a bad cable or 10/100 auto negotiation issues. Try known, working cables on a known working switch/hub port to see if the problem occurs. Try plugging it into another switch (preferably of different brand) if you have one. The 7960 cannot be forced to either 10 or 100mb operation, so if you do determine that it's not the cable or phone hardware, you'll have to either find a way to force 10 or 100mb operation on your switch or you'll have to get a new one... Once you get your phone hooked up to the network without problems, then you can start looking at the software issues.. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX pauses
Matteo == Brancaleoni Matteo [EMAIL PROTECTED] writes: Matteo turn off jitterbuffer in both servers. aka jitterbuffer=no in Matteo iax.conf Matteo jitterbuffer, unfortunately, is buggy and don't work as Matteo expected. Interesting -- this has indeed helped and the quality is better, too! But doesn't this mean I'm in trouble whenever the network decides to order packets around? --J. Matteo Il mer, 2003-07-16 alle 20:45, Jan Rychter ha scritto: Hi, I'm running asterisk in the following setup Phone - WX100USB - * - Internet - * - WX100P - PSTN The two Asterisks talk to each other via IAX2 and use GSM for voice. This seems to work quite well except for occasional pauses in voice transmission. These seem to occur in _one_ direction only (when I'm on the phone, I can't hear the person that I called via the PSTN), last several seconds (as in one to five seconds) and are unrelated to network connectivity (a ping in another window runs just fine all the time). What could be the cause? What else could I do to help hunt down that bug? --J. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] grandstream sip phone
Dlink has the dhp-90 (currently in limited release like Grandstream) for $60-70. It doesn;t have a digital display- but it works flawlessly. There are a few others- you just need to look around... -GSR - Original Message - From: marrandy [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 16, 2003 10:02 PM Subject: Re: [Asterisk-Users] grandstream sip phone On Wednesday 16 July 2003 03:52 pm, Greg Renouf wrote: Grandstream can improve the quality of their 'user interface' (many others have already accomplished this goal,) I can see very few situations where the $10-20 cost saving will make the quality sacrifice worthwhile. What other phones are in the $95-$105 range ??? Regards...Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX pauses
Il mer, 2003-07-16 alle 23:04, Jan Rychter ha scritto: Matteo == Brancaleoni Matteo [EMAIL PROTECTED] writes: Matteo turn off jitterbuffer in both servers. aka jitterbuffer=no in Matteo iax.conf Matteo jitterbuffer, unfortunately, is buggy and don't work as Matteo expected. Interesting -- this has indeed helped and the quality is better, too! yep. jitterbuffer must be fixed in iax(2) But doesn't this mean I'm in trouble whenever the network decides to order packets around? no. jitterbuffer isn't used to get the right packet order, but for assuring that each packet 'hit the phone' at the right time delta. Quick example: if my sample size is 20ms , I must receive a packet every 20ms, in order to have a good play. but network isn't perfect , so 2nd packet arrives 22ms, the 3rd 18 ms and so on... jitterbuffer is a buffer used to get some packets, buffer them and play it at the right delta time, in our example 20ms. Infact the real name for jitterbuffer is Adaptive Voice Playout Buffer . adaptative means that the software could increase/decrease the buffer depending on the delta time of packets. More is the delta differences, more is the buffer. If the delta diffs reduce, the buffer shrink . be aware that a jitterbuffer adds latency, so why is preferable to have it 'adaptive'. That's a very *light* and quick explanation , if you wanna learn more, google with 'Adaptive Voice Playout Buffer' ;) matteo -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39.02.70633354 - ext 911 IAX(2): [EMAIL PROTECTED] - ext 911 or tel:17005662458 - ext 911 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FXS and PBX Integration
Hi All, I got a doubt about something I want to do with asterisk. I have this office (site a) with only a Panasonic analog PBX and another office (site b) with an Asterisk Box with an ADIT 600 . I want to interconnect both via IAX. Is it possible to put a new asterisk box in site a without the channel bank and put a card (FXS or FXO???) and connect it to the pbx as a CO line ? What kind of card do I need a FXS or an FXO card? Regards, Iván Aponte -- Iván Aponte email: [EMAIL PROTECTED] Office: +58(212)9524620 Mobile: +58(414)2774713 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXS and PBX Integration
It sounds like you want to use the IAX to provide dialtone to the Panasonic PBX? You'd use FXS cards in the asterisk box to provide signal into a CO port on the Panasonic. On Wed, 16 Jul 2003, Iván Aponte wrote: Hi All, I got a doubt about something I want to do with asterisk. I have this office (site a) with only a Panasonic analog PBX and another office (site b) with an Asterisk Box with an ADIT 600 . I want to interconnect both via IAX. Is it possible to put a new asterisk box in site a without the channel bank and put a card (FXS or FXO???) and connect it to the pbx as a CO line ? What kind of card do I need a FXS or an FXO card? Regards, Iván Aponte -- Iván Aponte email: [EMAIL PROTECTED] Office: +58(212)9524620 Mobile: +58(414)2774713 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Segmentation fault with chan_oh323
Hi, I'm trying to interconnect sip and h323 endpoints using asterisk and asterisk crashes with segmentation fault whenever h323 connection needs to be established. It registers with gatekeeper ok though. Here are the symptoms. If the call initiated by SIP device, asterisk replies to it Trying and then silently crashes (it launched as asterisk -cd). In debug log I can see the following: Jul 16 18:11:52 DEBUG[196621]: File pbx.c, Line 1123 (pbx_extension_helper): Launching 'Dial' Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line 1393 (oh323_request): In oh323_request. Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line 1394 (oh323_request): type=oh323, format=4, data=phone number. Jul 16 18:11:52 DEBUG[196621]: File chan_oh323.c, Line 1440 (oh323_request): Created new call structure 0 (2428 bytes). That's it. If the call initiated by H323 device, then I see *CLI WrapH323Connection::WrapH323Connection: WrapH323Connection created. Segmentation fault and debug log shows: Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 2141 (init_h323_connection): In init_h323_connection... Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 2180 (init_h323_connection): Created new call structure 0 (2428 bytes). Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1527 (copy_call_details): --- CALL DETAILS --- Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1528 (copy_call_details): call_token = ip$192.168.0.227:5018/92 Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1529 (copy_call_details): call_source_alias = tnt [192.168.0.227] Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1530 (copy_call_details): call_dest_alias = 12125551234 12125551234 ip$192.168.0.70:1720 Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1531 (copy_call_details): call_source_e164 = phone number Jul 16 18:33:12 DEBUG[196621]: File chan_oh323.c, Line 1532 (copy_call_details): call_dest_e164 = 12125551234 That's it. And gatekeeper log shows that after normal ARQ-ACF exchange originating device immediately sent DRQ. If anybody knows a reason for this (and the way to fix it of course ;)), I'd appreciate if you let me know. If you need any additional info to troubleshoot it, let me know too. Thank a lot. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice Modem + Soundcard Driver
- Original Message - From: Anthony Wood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, June 03, 2003 6:04 PM Subject: Re: [Asterisk-Users] Voice Modem + Soundcard Driver On Tue, Jun 03, 2003 at 05:47:54PM +1000, Mathew Frank wrote: Woody wrote: The problem with using Voice Modems is that they fall into two categories: 1) Hardware Modems which only have half-duplex transmission of voice 2) Soft/Win/Lin modems which are proprietry and don't have asterisk drivers or 3 - full duplex real voice modems such as produced by Banksia in Sydney which have been available for years for IVR systems for which only a single-duplex channel is available in Asterisk. Would it be hard to code a full duplex asterisk driver? Or is it a kernel driver issue? Oh, I see that you can download linux drivers from Banksia... It is purely as issue with the Asterisk Channel Driver. The Banksia kernel driver is full-duplex. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Back-to-back connected boards load test
Title: Back-to-back connected boards load test Hi, Two asterisk boards in two asterisk boxes are connected back-to-back with crossover cables. T400P (box1, pri_net) creates a flow of calls, TE410P (box2, pri_cpe) takes them. Cables are OK. Boxes are powerful. SMP is in place. Interrupts are distributed among CPUs. There are 2 TE410Ps in the box2. Spans 1-4 in box1 are connected to spans 5-8 in box2. The carrier T1 is in span 1 of box2 and provides clocking. All we want is to be able to have a consistent flow of 92 calls. All we can do is 75-80 calls. There are D-Channels downs and ups on the spans where new calls are sent. There are frame rejections and retransmissions. There are read failures on D-Channel fds with 'Unknown error 500'. There are PRI events 6 and 8 (HDLC-related). It gets worse as the total amount of calls goes up. The log (from asterisk start to the test finish is attached). Would greatly appreciate any advice on how to fight this. Thank you. Alex Zarubin Webley Systems, Inc. p-c.gz p-c.gz Description: Binary data
[Asterisk-Users] Question on peer to peer config
I want to setup 2 asterisk boxes in 2 different statesto make long distances calls.. I know I need to get a TDM400P 1 port FXS and a X100P I think that I can use IAX to connect them over the internet I want to be able to pick up the phone in state one, and have it pick up the line attached to the X100P in the other state. I know I might have to enter a extention to get to the x100p. I also want any calls comming in over the X100p to ring the phone in the other state. Has this been done by anyonedoes anyone feel like sharing configs? Thanks, Michael Hess Uc9 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems getting 7960's to play nice withAsterisk
Ok so no more ethernet disconnected errors. However, I still am getting the same voice mail errors. (BTW, don't plug the cable into the wrong port on the back... sigh)... The phone starts the call and then immediately disconnects. I've included the * console, and tethereal dumps. Any Ideas? Here is what I see on the * side of things: *CLI -- Registered SIP '2000' at 192.168.3.126 port 5060 expires 120 -- Executing Dial(SIP/2000-1b5f, SIP/2000|20) in new stack -- Called 2000 -- Got SIP response 488 Not Acceptable Here back from 192.168.3.126 == No one is available to answer at this time -- Executing VoiceMail(SIP/2000-1b5f, u2000) in new stack == Parsing '/etc/asterisk/voicemail.conf': Found -- Playing 'vm/2000/unavail' == Spawn extension (from-sip, 2000, 2) exited non-zero on 'SIP/2000-1b5f' Tethereal dump: Capturing on eth0 0.00 192.168.3.126 - 192.168.3.5 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 0.002663 192.168.3.5 - 192.168.3.126 SIP Status: 407 Proxy Authentication Required 0.065339 192.168.3.126 - 192.168.3.5 SIP Request: ACK sip:[EMAIL PROTECTED] 0.116006 192.168.3.126 - 192.168.3.5 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 0.116629 192.168.3.5 - 192.168.3.126 SIP Status: 100 Trying 0.117865 192.168.3.5 - 192.168.3.126 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 0.287018 192.168.3.126 - 192.168.3.5 SIP Status: 488 Not Acceptable Here 0.287392 192.168.3.5 - 192.168.3.126 SIP Request: ACK sip:[EMAIL PROTECTED] 0.287785 192.168.3.5 - 192.168.3.126 SIP/SDP Status: 200 OK, with session description 0.451552 192.168.3.126 - 192.168.3.5 SIP Request: ACK sip:[EMAIL PROTECTED]:5060 0.475174 192.168.3.126 - 192.168.3.5 SIP Request: BYE sip:[EMAIL PROTECTED]:5060 0.475496 192.168.3.5 - 192.168.3.126 SIP Status: 200 OK On Wed, 16 Jul 2003 [EMAIL PROTECTED] wrote: This phone was working with the same cable/switch when used with a Skinny configuration. But I'll try your suggestion out just to make sure when I get home tonight... -Steve On Wed, 16 Jul 2003, John Laur wrote: Other things: The phone constantly says Ethernet Disconnected. Even though it tftp's configs and registers with the proxy. Something is wrong with either the phone hardware itself, the network port on the hub or switch it is attached to, the ethernet cable it is connected with, or the 10/100 autonegotiation between the phone and the switch. Any other problems you are having are quite likely due to these network problems, and so you should fix this problem before you attempt to debug any of the others. The Ethernet Disconnected message on the phones is the equivalent of having no link light on your NIC. Standard network connection debugging applies. Some tips: This problem is likely due to a bad cable or 10/100 auto negotiation issues. Try known, working cables on a known working switch/hub port to see if the problem occurs. Try plugging it into another switch (preferably of different brand) if you have one. The 7960 cannot be forced to either 10 or 100mb operation, so if you do determine that it's not the cable or phone hardware, you'll have to either find a way to force 10 or 100mb operation on your switch or you'll have to get a new one... Once you get your phone hooked up to the network without problems, then you can start looking at the software issues.. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problems getting 7960's to play nice withAsterisk
Figured it out... sip.cnf: change: allow=all to: allow=ulaw ta da! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Pickup
Hi, I have been trying to workout how to use the call pickup. So Far, I have the following in zapata.conf [channels] signalling = fxo_ks context = local pickupgroup=1 callgroup=1 channel = 1-3 When I dial *8# all I hear is busy tone. What have I missed? thanks Jay. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
